Third-party call control allows a non-endpoint entity to originate and manage a call between other entities. Early telephony networks relied on the premise that all calls begin at an endpoint that provided the signaling and interface necessary to make the call. Even within more modern telephony networks, basic network operations required that a call originate at an endpoint possessing capabilities for media and signaling. Consequently, third-party call control was not supported in prior public switched telephone networks (PSTNs). As PSTN standards advanced, third-party call control mechanisms were introduced, but few third-party call control solutions were widely implemented. Although the premise that all calls begin at an endpoint capable of providing signaling and appropriate interfaces was sufficient for plain old telephone services (POTS), the premise potentially restricted the evolution of enhanced services and data interfaces.
As softswitches were introduced, media and signaling could effectively be separated. A softswitch is an application program interface (API) used to connect a traditional PSTN to a voice over internet protocol (VOIP) product. The softswitch linked the PSTN to IP networks and managed multiple signaling types.
Today, multimedia networks are based on Internet protocols and, therefore, allow for the separation of media and signaling and the separation of applications from the signaling and media aspects of a communications session. Consequently, third-party application servers may effectively manage the attributes of a call between participants. The session initiation protocol (SIP) was created, among other reasons, to support the creation of real-time communication sessions in IP networks. Together with the session description protocol (SDP), the session initiation protocol (SIP) may effectively separate the transfer of media and signaling within a communication session.
Although SIP provides a mechanism for implementing third party call control, several approaches have been introduced to enable third-party call control, but each approach possesses significant disadvantages. The most common approach is to develop SIP third-party call control that closely resembles the current computer-telephony-integration (CTI) model. In this approach, a back-to-back user agent controls and bridges the multiple call legs to other user agents. Unfortunately, the end-to-end semantics of SIP are broken, because each user agent views itself in session with the back-to-back user agent instead of the other endpoint. The third-party controller must explicitly be aware of the back-to-back user agent for this approach to be effective.
In another SIP approach, third-party call control is modeled using a peer-to-peer configuration. This SIP approach eliminates the need for a back-to-back user agent, but increases the complexity of the type of SIP primitives an endpoint must process. Although such an approach may be desirable, no such successful implementation is known to have existed in the past.
Accordingly, there is a need in the industry for systems and methods for controlling and monitoring communication devices through third-party call control using a peer-to-peer configuration with SIP.
Therefore, there is a need in the industry for systems and methods of controlling and monitoring communication devices that provide third-party call control and that overcome the complexities involved with implementing a third-party call control using a peer-to-peer configuration with SIP.